/*
 *  Copyright 2019 The WebRTC Project Authors. All rights reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#ifndef MODULES_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_
#define MODULES_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_

#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"

namespace webrtc {

// Represents fields and derived information received in RTCP report block
// attached to RTCP sender report or RTCP receiver report, as described in
// https://www.rfc-editor.org/rfc/rfc3550#section-6.4.1
class ReportBlockData {
 public:
  ReportBlockData() = default;

  ReportBlockData(const ReportBlockData&) = default;
  ReportBlockData& operator=(const ReportBlockData&) = default;

  // The SSRC identifier for the originator of this report block,
  // i.e. remote receiver of the RTP stream.
  uint32_t sender_ssrc() const { return report_block_.sender_ssrc; }

  // The SSRC identifier of the source to which the information in this
  // reception report block pertains, i.e. local sender of the RTP stream.
  uint32_t source_ssrc() const { return report_block_.source_ssrc; }

  // The fraction of RTP data packets from 'source_ssrc()' lost since the
  // previous report block was sent.
  // Fraction loss in range [0.0, 1.0].
  float fraction_lost() const { return fraction_lost_raw() / 256.0; }

  // Fraction loss as was written in the raw packet: range is [0, 255] where 0
  // represents no loss, and 255 represents 99.6% loss (255/256 * 100%).
  uint8_t fraction_lost_raw() const { return report_block_.fraction_lost; }

  // The total number of RTP data packets from 'source_ssrc()' that have been
  // lost since the beginning of reception.  This number is defined to be the
  // number of packets expected less the number of packets actually received,
  // where the number of packets received includes any which are late or
  // duplicates. Thus, packets that arrive late are not counted as lost, and the
  // loss may be negative if there are duplicates.
  int cumulative_lost() const { return report_block_.packets_lost; }

  // The low 16 bits contain the highest sequence number received in an RTP data
  // packet from 'source_ssrc()', and the most significant 16 bits extend that
  // sequence number with the corresponding count of sequence number cycles.
  uint32_t extended_highest_sequence_number() const {
    return report_block_.extended_highest_sequence_number;
  }

  // An estimate of the statistical variance of the RTP data packet interarrival
  // time, measured in RTP timestamp units. The interarrival jitter J is defined
  // to be the mean deviation (smoothed absolute value) of the difference D in
  // packet spacing at the receiver compared to the sender for a pair of
  // packets.
  uint32_t jitter() const { return report_block_.jitter; }

  // Jitter converted to common time units.
  TimeDelta jitter(int rtp_clock_rate_hz) const;

  // TODO(danilchap): Deprecate in favor of using ReportBlockData accessors
  // directly.
  const RTCPReportBlock& report_block() const { return report_block_; }

  [[deprecated]] int64_t report_block_timestamp_utc_us() const {
    return report_block_timestamp_utc_.us();
  }
  [[deprecated]] int64_t last_rtt_ms() const { return last_rtt_.ms(); }
  [[deprecated]] int64_t min_rtt_ms() const { return min_rtt_.ms(); }
  [[deprecated]] int64_t max_rtt_ms() const { return max_rtt_.ms(); }
  [[deprecated]] int64_t sum_rtt_ms() const { return sum_rtt_.ms(); }

  // Time in utc epoch (Jan 1st, 1970) the report block was received.
  Timestamp report_block_timestamp_utc() const {
    return report_block_timestamp_utc_;
  }

  // Round Trip Time measurments for given (sender_ssrc, source_ssrc) pair.
  // Min, max, sum, number of measurements are since beginning of the call.
  TimeDelta last_rtt() const { return last_rtt_; }
  TimeDelta min_rtt() const { return min_rtt_; }
  TimeDelta max_rtt() const { return max_rtt_; }
  TimeDelta sum_rtts() const { return sum_rtt_; }
  size_t num_rtts() const { return num_rtts_; }
  bool has_rtt() const { return num_rtts_ != 0; }

  void SetReportBlock(uint32_t sender_ssrc,
                      const rtcp::ReportBlock& report_block,
                      Timestamp report_block_timestamp_utc);
  void AddRoundTripTimeSample(TimeDelta rtt);

 private:
  RTCPReportBlock report_block_;
  Timestamp report_block_timestamp_utc_ = Timestamp::Zero();
  TimeDelta last_rtt_ = TimeDelta::Zero();
  TimeDelta min_rtt_ = TimeDelta::Zero();
  TimeDelta max_rtt_ = TimeDelta::Zero();
  TimeDelta sum_rtt_ = TimeDelta::Zero();
  size_t num_rtts_ = 0;
};

class ReportBlockDataObserver {
 public:
  virtual ~ReportBlockDataObserver() = default;

  virtual void OnReportBlockDataUpdated(ReportBlockData report_block_data) = 0;
};

}  // namespace webrtc

#endif  // MODULES_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_